similar to: jitterbuffers - suggested settings

Hi there, I support a large number of enterprise users who contractually must connect to our support center via a 4G VOIP connection. I simply want to be able to auto detect all poor quality calls in realtme (as they are being made), play a message and drop the call - without user intervention. All decent call quality calls will be allowed through - to be handled by support staff. Its a

We're looking at implementing Asterisk in our department in the near future, we're looking at anywhere from 15-25 extensions. The machine we were looking at running this on was a Quad Xeon 450mhz (2MB L2 Cache) w/ 1GB of ram. I've heard bad things about running Asterisk on SMP machines? Would we be running into any performance issues with this machine? Tim Jackson Network Engineer

Steve Kann has developed a new jitterbuffer for IAX2, that hopefully will be integrated into Asterisk v1.1 soon, to be part of the 1.2 stable relase. Zoa and his bulgarian team is porting this buffer to SIP/RTP, but needs support in the form of funding in order to take the time to test this out and complete it in time. Please paypal your contribution to sponsor@astertest.com today. Every

In my series of articles about Asterisk 1.4, I've now arrived to the new jitter buffer that enhances voice quality for those of you using Asterisk as a PSTN gateway. Please read http://www.voip-forum.com/category/asterisk/asterisk14/ /O

Hi, 1. Has anyone got any success when send a TIFF file form one zoiper softphone to another ? I tried using Zoiper 2.18 free edition in windows but I'm seeing 415 Unsupported media replies. 2. Here (http://www.voipinfo.org/wiki/view/Asterisk+T.38), you can read : "Also, try using: t38_udptl=yes t38pt_rtp=no t38pt_tcp=no ... in the general section of the sip.conf and under the VoIP

Hi, I'd like to test Asterisk performance under more concurrent sip calls. I use Sipp, but do get "Unexpected message for Call-ID ...", so I wonder if anyone is using sipp succesfully with Asterisk and is willing to share more info about his solution ... Any other convenient way to load test Asterisk ? Is sipp the right tool ? Thanks in advance, regards, Rob. sipp: The

Hello Is it correct that IAX2 uses more CPU, than SIP? Also, can it be true that IAX2 is much more sensitive against high CPU loads? Also, does Asterisk support and use multiprocessor architectures, such as Xeon? ? Regards Jon -- No virus found in this outgoing message. Checked by AVG Free Edition. Version: 7.1.394 / Virus Database: 268.8.3/362 - Release Date: 12-06-2006

> Yes. Jean-Marc has made the API more similar. > > Jean-Marc: Have you looked at the API we have for the > asterisk/iaxclient jitterbuffer? Just did. > It's pretty close to what you have now -- the major difference is that > your jb still assumes it can "own" the data passed in -- it copies it, > and it destroys it at will. With the API I put together,

Hello! I am going to use the JITTERBUFFER function in a SIP (and local channels) only setup, but have some questions of how to use it: 1. Do I need to activate jbenable in sip.conf? Or is it enough to call the JITTERBUFFER function? 2. What is the preferred way to invoke this function? Say I have channel A which is not in need of buffering, while channel B do need it. If A

Does anyone test Asterisk 1.2.X + bristuff-0.3.X and TDM card? We can't get it to work. -David -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Julian J. M. Sent: Friday, March 31, 2006 1:44 AM To: Chris Earle; Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Re: BRI

What is the consensus on using the 1.4 jitterbuffer? Do most people enable it? We have a "PSTN" server that has our RBS T1 trunks in a central location, then have clients that connect via SIP to us for access to those trunks. Most of them are just fine, but lately we have a handful that are having latency and jitter issues. I am hesitant to just turn on the jitter buffer in

WTF is a jitterbuffer? Sent from my Verizon Wireless 4G LTE smartphone -------- Original message -------- From: Matthew Jordan <mjordan at digium.com> Date: 01/29/2015 10:41 AM (GMT-05:00) To: Asterisk Users Mailing List - Non-Commercial Discussion <asterisk-users at lists.digium.com> Subject: Re: [asterisk-users] JITTERBUFFER function On Thu, Jan 29, 2015 at 4:56 AM,

> We just return a frame with the return value JB_DROP, which tells the > caller to drop this frame, and call jb_get again. > > When the caller is done with the jitterbuffer, it calls jb_getall() > repeatedly, until it's empty, and then it can discard all the frames. Hmm, looks a bit error-prone to me. Especially considering I still have to explain that "no, you

I find that DTMF does not work reliably if jitterbuffer=on for certain IAX providers. For instance, DTMF tones are missed entirely about half the time when I dial into an exgn.net account. However, it always works fine for an unlimitel.ca account. Someone else has seen this too: http://bugs.digium.com/view.php?id=6011 Can anyone suggest a workaround (other than jitterbuffer=off)? - Mike

In iax.conf there is option jitterbuffer how about sip protocol ? Are jitterbuffer can configure in sip.conf ? Thanks, for your share

I am trying to work out IAX <--> IAX communications with my * box. I have a registration with iaxtel and I thought I would start there for my learning. I am able to call the number for Digium's support line (700-428-6000), but the sound is horribly chopping. Some reading revealed the jitterbuffer settings, so I enabled them in iax.conf. I have the following now: ; Inter-Asterisk

Does anyone know if the tutorial materials from Atricon 2004 are available for download anywhere? I'm particularly interested in Joachim Vanheuverzwijn's Performance and Scalability tutorial slides (Asterisk - building your system for performance and scalability). Thanks!

I'm having a problem with intersite calls over IAX2 being abruptly terminated. Nothing odd shows in any of the logs for Asterisk or the host. The only think I can think it might be is a lag-spike on the site to site connection. How sensitive is IAX2 to lost frames, lag spikes or large variations in jitter with the GSM codec and: bandwidth=low jitterbuffer=no trunkfreq=100 ; Raised from

> Perhaps, but then you need to assume that the jitterbuffer can just > throw away the data, and that limits how you can use it. In object- > oriented terms, you might want to pass objects to the JB, and then > call a destructor on them. In C terms, you may want to allocate > frames via malloc(), and then call free() on them later. You might > want to pass in

> > 1. Do I need to activate jbenable in sip.conf? Or is it enough to call > > the JITTERBUFFER function? > > You only need to use the JITTERBUFFER function. > > The jbenable option will enable a jitter buffer on every channel > created for that peer (or, if global, for every peer in the system). > Depending on the version of Asterisk, it will also place the